As it is generally known, packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. Various network conditions may cause packet loss, including network congestion.
Packet loss is sometimes addressed by retransmission of lost data. For example, the Transmission Control Protocol (TCP) uses retransmission to guarantee correct delivery of data even when packets are lost in transit.
Another approach to handling packet loss is to transmit data redundantly, so that the receiving device can reconstruct the lost data without requesting retransmission. Redundant data transmission techniques are generally referred to as forward error correction (FEC). FEC enables correction of some errors at a receiver without retransmission of data, but requires a higher channel bandwidth for data transmission. Examples of FEC include techniques that apply Reed-Solomon codes, Hamming codes, Golay codes, Reed-Muller codes, Turbo codes, and Low Density Parity Check (LDPC) codes.
Failure to address packet loss during network-based audio data communications may result in reduced quality of the audio output by receiving devices that experience the packet loss. However, existing techniques for handling packet loss, such as packet retransmission and/or FEC, introduce delays that may reduce the ability of users, such as participants in an online meeting, to hold real-time conversations over the network. In this way, some techniques for addressing network packet loss may reduce the “conversational quality” of audio communications being provided over a network.